[asterisk-users] Set Call type in dial plan

[Sammy Govind]

Hi,
Sorry for late reply. Hope you've already found out something about it.

What version of asterisk you are using, that function for choosing
inbound/outbound call leg codecs is for newer versions of asterisk.
See these pages:
http://www.voip-info.org/wiki/view/Asterisk+variables
https://issues.asterisk.org/view.php?id=13243

Regards,
Sammy


On Tue, Jan 3, 2012 at 2:31 PM, Faraj Khasib <fkhasib at iconnecths.com> wrote:

> thats excatly what I want, can u plz give me the command, I want to choose
> only ulow
> ________________________________________
> From: asterisk-users-bounces at lists.digium.com [
> asterisk-users-bounces at lists.digium.com] On Behalf Of Sammy Govind [
> govoiper at gmail.com]
> Sent: Tuesday, January 03, 2012 3:26 AM
> To: Asterisk Users Mailing List - Non-Commercial Discussion
> Subject: Re: [asterisk-users] Set Call type in dial plan
>
> Hi,
>
> For such call you just need to select the outbound codec before the dial()
> app.
>
> choose the audio-only codecs and thus no video codec strings will be
> exchanged in that call.
>
> --
> Regards,
> Sammy
>
> On Tue, Jan 3, 2012 at 1:54 PM, Faraj Khasib <fkhasib at iconnecths.com
> <mailto:fkhasib at iconnecths.com>> wrote:
> this is what my SIP Invite message when I make Video call
>
> INVITE sip:6500 at 192.168.21.102<mailto:sip%3A6500 at 192.168.21.102> SIP/2.0
> Via: SIP/2.0/UDP 192.168.21.193:52933;branch=z9hG4bK1943005978;rport
> From: <sip:6097 at 192.168.21.102<mailto:sip%3A6097 at 192.168.21.102
> >>;tag=1857098215
> To: <sip:6500 at 192.168.21.102<mailto:sip%3A6500 at 192.168.21.102>>
> Contact: <sip:6097 at 192.168.21.193:52933
> ;transport=udp>;+g.oma.sip-im;language="en,fr";+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
> Call-ID: b9453704-d76a-b8ce-3247-c999abff7395
> CSeq: 324677463 INVITE
> Content-Type: application/sdp
> Content-Length: 588
> Max-Forwards: 70
> Route: <sip:192.168.21.102:5060;lr;transport=udp>
> Accept-Contact:
> *;+g.3gpp.icsi-ref="urn%3Aurn-7%3A3gpp-service.ims.icsi.mmtel"
> P-Preferred-Service: urn:urn-7:3gpp-service.ims.icsi.mmtel
> Allow: INVITE, ACK, CANCEL, BYE, MESSAGE, OPTIONS, NOTIFY, PRACK, UPDATE,
> REFER
> Privacy: none
> P-Access-Network-Info: ADSL;utran-cell-id-3gpp=00000000
> User-Agent: Medcor
> Supported: 100rel
>
> v=0
> o=doubango 1983 678901 IN IP4 192.168.21.193
> s=-
> c=IN IP4 192.168.21.193
> t=0 0
> m=audio 36372 RTP/AVP 8 0 9 101
> a=ptime:20
> a=rtpmap:8 PCMA/8000/1
> a=rtpmap:0 PCMU/8000/1
> a=rtpmap:9 G722/8000/1
> a=rtpmap:101 telephone-event/8000/1
> a=fmtp:101 0-15
> m=video 59296 RTP/AVP 125 106 121 103
> a=rtpmap:125 VP8/90000
> a=fmtp:125 CIF=2;QCIF=2;SQCIF=2
> a=rtpmap:106 H264/90000
> a=fmtp:106 profile-level-id=42e01e; packetization-mode=1; max-br=452;
> max-mbps=11880
> a=rtpmap:121 MP4V-ES/90000
> a=fmtp:121 profile-level-id=3
> a=rtpmap:103 H263-1998/90000
> a=fmtp:103 CIF=2;QCIF=2;SQCIF=2
>
> when I make Audio call requests I dont have the video part .... but at
> receiver since two clients can make video call they have Asterisks adds the
> Video Part in request sent to receiver,I dont want that part added , how I
> can delete it ?
> --
> _____________________________________________________________________
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>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>               http://www.asterisk.org/hello
>
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